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Personal audio systems have been developed based on various approaches with the goal of synthesizing an isolated sound zone that avoids disturbing others in different locations. In this work, a near-field solution for a neckband headset using three loudspeakers positioned close to each ear is proposed. In particular, it is an attempt to derive a simple multichannel filter for reducing the computational cost in mobile devices. Unlike super-directive beamforming techniques, the controlled radiation pattern is not highly directional but can boost the near-field sound, thereby providing an extra sound level difference between the listener's ear locations and far-field surrounding areas. For this purpose, a multichannel filter is designed using a conventional pressure matching technique for reproducing a target signal at the ear location while suppressing sound radiation to a far-field. It is shown that the optimal filter weights can be successfully approximated in the form of a simple broadside differential array pattern. The simplified filter structure can be realized using only two second-order infinite impulse response filters for driving the middle and two side loudspeakers. Through various simulations and experiments, it is demonstrated that the proposed solution can effectively realize a personal audio system with a minimal loss of sound isolation performance.

Measurement of noise exposure from communication headsets is challenging due to the need for specialized equipment, methods, and training. Canadian standard Z107.56-13 [(2013). Measurement of Noise Exposure (Canadian Standards Association, Mississauga, Canada)] introduced a calculation procedure to promote a simpler method that stakeholders in hearing loss prevention could readily apply using widely accessible sound level equipment. The original procedure specified a fixed signal-to-noise ratio (SNR) of 15 dB above the protected background noise when estimating the speech listening level through the headset communication channel. The relationship between background noise level, noise type, headset noise reduction (NR), and speech listening level is revisited in this study. In a noise simulation room, 24 participants were asked to adjust the headset volume while listening to speech and executing a visual reaction task. Results indicate that the growth in speech listening levels with noise is not adequately represented by a fixed SNR, and that one-sided listening increases speech levels by about 5-7 dB in quiet and in noise compared to two-sided listening. Moreover, use of an octave-band procedure with derating to estimate headset NR best captured the speech listening level data. A revised calculation procedure based on linear regression modeling is described with parameters adjusted separately for one-sided and two-sided headsets.

Estimating the basic acoustic parameters of conversational speech in noisy real-world conditions has been an elusive task in hearing research. Nevertheless, these data are essential ingredients for speech intelligibility tests and fitting rules for hearing aids. Previous surveys did not provide clear methodology for their acoustic measurements and setups, were opaque about their samples, or did not control for distance between the talker and listener, even though people are known to adapt their distance in noisy conversations. In the present study, conversations were elicited between pairs of people by asking them to play a collaborative game that required them to communicate. While performing this task, the subjects listened to binaural recordings of different everyday scenes, which were presented to them at their original sound pressure level (SPL) via highly open headphones. Their voices were recorded separately using calibrated headset microphones. The subjects were seated inside an anechoic chamber at 1 and 0.5 m distances. Precise estimates of realistic speech levels and signal-to-noise ratios (SNRs) were obtained for the different acoustic scenes, at broadband and third octave levels. It is shown that with acoustic background noise at above approximately 69 dB SPL at 1 m distance, or 75 dB SPL at 0.5 m, the average SNR can become negative. It is shown through interpolation of the two conditions that if the conversation partners would have been allowed to optimize their positions by moving closer to each other, then positive SNRs should be only observed above 75 dB SPL. The implications of the results on speech tests and hearing aid fitting rules are discussed.

Virtual and Augmented Reality (VR and AR) are increasingly becoming a consumer commodity, where sound plays a central role, usually delivered through headphones. Headphones, however, do not allow direct interactions between users outside the virtual world: other solutions may be necessary. In this work, we present a different approach: a sound projector, which can deliver spatial sound at a distance by forming a beam of audible sound. Our device consists of a portable speaker and two metamaterial lenses of different focal length, arranged in a telescope. We test our device by delivering signals between 5 and 6 kHz to moving people, tracked by a camera. We benchmark its performance with measurements and a pilot study, with human participants. Finally, we connect our projector to a VR headset and show how it can be used to deliver messages in VR. In this work, we discuss the limitations and perspectives of similar metamaterial-based delivery systems.

The acoustic properties of a space influence how it is being perceived. However, less is known about the ways in which the acoustic environment elicit different emotional responses. The purpose of this paper is to investigate this question through the study of worship spaces. The Focus on such buildings in this study is due to the essential role of the acoustics in shaping the spiritual experience. Adopting a mixed-method approach, the researchers created virtual experiences of these buildings using 360-deg sound and visual recordings. The recordings were displayed for the subjects through a virtual reality headset. During the experiment, physiological signals of the participants and semantic descriptions of their experience were collected and analyzed to understand the emotional impact of the acoustic environment. The outcome of this study will allow exploring the relationship between acoustic characteristics of worship spaces and the human emotional experience. Therefore, it will provide a further understanding of the impact of the architectural design that determines the acoustic environment, which then contributes to improving the restorative qualities of such spaces and enhancing the spiritual experience.

The NIDCD has funded a number of projects to develop portable signal processing tools that enable real-time processing of the acoustic environment. The overarching goal is to provide a large group of researchers with the means to efficiently develop and evaluate novel signal processing schemes, individualized fitting procedures, and technical solutions and services for hearing apparatus such as hearing aids and assistive listening devices. We report here on a development done in the SBIR Phase II Project R44DC016247. This project builds on the software being concurrently developed in R01DC015429 to provide a complete portable and wearable software-hardware master hearing aid device needed for development of new solutions for assisted hearing. We will present and demonstrate the portable platform, currently in the Beta launch, that consists of a Cortex A8 based processing unit and a codec set able to support hearing aid architecture of up to 6 microphones and 4 speakers. It is currently accompanied by a binaural 2-microphone BTE hearing aid set, but will also support different headset form-factors of our partners. Additionally it features stereo line in and line out connections. The device can be remotely controlled with a smart phone.The NIDCD has funded a number of projects to develop portable signal processing tools that enable real-time processing of the acoustic environment. The overarching goal is to provide a large group of researchers with the means to efficiently develop and evaluate novel signal processing schemes, individualized fitting procedures, and technical solutions and services for hearing apparatus such as hearing aids and assistive listening devices. We report here on a development done in the SBIR Phase II Project R44DC016247. This project builds on the software being concurrently developed in R01DC015429 to provide a complete portable and wearable software-hardware master hearing aid device needed for development of new solutions for assisted hearing. We will present and demonstrate the portable platform, currently in the Beta launch, that consists of a Cortex A8 based processing unit and a codec set able to support hearing aid architecture of up to 6 microphones and 4 speakers. It is currently accompanied by a bi...

Simulations of sensorineural hearing loss (SNHL) and bilateral cochlear implantation (BCI) have been modeled successfully under static and non-real time conditions. This study performed two experiments testing the validity of a novel real-time SNHL/BCI simulation application for iOS using an in-ear binaural-recording headphone apparatus. The first experiment measured spatial release from masking (SRM) with normal hearing (NH), against headset apparatus simulations of NH, SNHL, and BCI using HINT sentences, speech shaped noise and forward masking. A one-sample t-test revealed significant differences between NH and simulated SNHL and BCI conditions showing reduced benefit from SRM. The second experiment employed noise bursts across nine frontal-plane loudspeakers and measured localization accuracy under NH and six frequency band BCI simulation conditions. Repeated measures two-way ANOVA and Cronbach’s Alpha suggested significantly reduced localization ability with BCI simulation. While further testing is needed, results here provide promising evidence that real-time binaural recording with low-latency processing and in-ear playback may be used to simulate SNHL and the BCI percept in NH listeners. The limitations and potential of this technology to expand the subject pool and expedite innovative testing are discussed.

Recreational and private pilots are regularly exposed to potentially hazardous noise levels during flight training sessions and recreational flying. For the first part of this study, light airplane noise was recorded during recurrent 1.5-hour flight training sessions. Measurements were made at the level of the pilot’s left and right shoulders (using dosimeters). Data demonstrated that changing inflight operations and power adjustments were associated with specific acoustic patterns. Noise levels (Leq) ranged from 76 dBA to 96 dBA during flight. The second part of this study examined the effect of the noise on subjective and objective measures of the pilot’s auditory function. Measurements were made before and immediately after flight training sessions. Pure-tone audiometric thresholds, middle-ear absorbance and click-evoked otoacoustic emissions did not show significant changes from pre- to post-flight. Although inflight noise levels are potentially hazardous, the use of an aviation headset acted to mitigate the risk of hazardous exposure. Future research directions include measurements in diverse aircraft and across longer flight durations.

Specialized standardized methods for the measurement of noise exposure from communication headsets or sound sources close to the ear include the Microphone in Real Ear and manikin techniques, as specified in ISO 11904-1/2. The 2013 version of Canadian standard Z107.56 introduced a simpler calculation method to increase accessibility to communication headset exposure assessments for the widest range of stakeholders in hearing loss prevention. The calculation method only requires general and widely accessible sound measurement equipment and basic computational steps that account for the main determinants of exposure such as the background noise around the user, the sound attenuation of the communication headset, and the expected communication duration and effective listening signal-to-noise ratio. This paper reviews recent research on the effects of the spectral and temporal characteristics of the background noise and the headset configuration on the speech listening level. Results indicate that the listeni...

Active noise-reducing headsets that employ analog feedback control and provide good broadband attenuation are commercially available for a wide range of applications. Recent studies have explored the integration of an adaptive digital feedforward controller with the analog feedback controller to provide additional attenuation of periodic noise components. This paper presents an experimental study of such a combined control system, but with both feedback- and feedforward controllers attenuating broadband noise. Good performance is demonstrated in a reverberant sound field, while under direct sound-field conditions the attenuation performance of the feedforward controller is shown to be dependent on head position. The paper concludes with an analysis of the forward path delay showing how the passive attenuation mechanism improves broadband performance.

An adaptive leaky normalized least-mean-square (NLMS) algorithm has been developed to optimize stability and performance of active noise cancellation systems. The research addresses LMS filter performance issues related to insufficient excitation, nonstationary noise fields, and time-varying signal-to-noise ratio. The adaptive leaky NLMS algorithm is based on a Lyapunov tuning approach in which three candidate algorithms, each of which is a function of the instantaneous measured reference input, measurement noise variance, and filter length, are shown to provide varying degrees of tradeoff between stability and noise reduction performance. Each algorithm is evaluated experimentally for reduction of low frequency noise in communication headsets, and stability and noise reduction performance are compared with that of traditional NLMS and fixed-leakage NLMS algorithms. Acoustic measurements are made in a specially designed acoustic test cell which is based on the original work of Ryan et al. ["Enclosure for low frequency assessment of active noise reducing circumaural headsets and hearing protection," Can. Acoust. 21, 19-20 (1993)] and which provides a highly controlled and uniform acoustic environment. The stability and performance of the active noise reduction system, including a prototype communication headset, are investigated for a variety of noise sources ranging from stationary tonal noise to highly nonstationary measured F-16 aircraft noise over a 20 dB dynamic range. Results demonstrate significant improvements in stability of Lyapunov-tuned LMS algorithms over traditional leaky or nonleaky normalized algorithms, while providing noise reduction performance equivalent to that of the NLMS algorithm for idealized noise fields.

It is well-acknowledged that soundscape investigations must be carried out in the original context. In order to investigate the influence of acoustic cues on the perceptual measurements, binaural recordings were performed during soundwalks in the center of Gothenburg (Sweden) using a calibrated binaural headset. The participants were 20 students from the Chalmers Technical University. The evaluations were carried out in-situ regarding, e.g., loudness, appropriateness, pleasantness, and eventfulness at eight different outdoor gathering spaces. The group of students was divided into four groups. Two groups went from location 1 to 8, separated in time by 15 minutes. The other two groups went from location 8 to 1. At all eight locations each group recorded the acoustic environment (overall 32 recordings). The laboratory assessments were performed with the same students the next day using four recordings measured at four different locations. First, the evaluations were performed simultaneously in the classroom...

This research examines the test parameters in two methods proposed by an ANSI/ASA working group for measurement of auditory localization with tactical communications and hearing protection systems (TCAPS). The localization task consisted of responding to pink noise bursts (0.250s and 7000s). Participants responded by pointing to the perceived source with a laser pointer mounted on an instrumented chair. 12 participants completed the study—half of the trials while wearing no headset and half while wearing an active Invisio X5 headset. Method 1 used four pairs of loudspeakers, separated by 8°, 10°, or 12°, and the listener was tested in two orientations (0° and 45°). Intended for use in untreated acoustic environments, Method 1 provides only a coarse estimate of localization accuracy and proportion of front/back errors. Method 2 used 36 equally spaced loudspeakers, a sound treated environment, and a chair instrumented with a laser pointer. Performance is compared as a function of separation between loudspea...

An adaptive, delayless, subband feed-forward control structure is employed to improve the speech signal-to-noise ratio (SNR) in the communication channel of a circumaural headset/hearing protector (HPD) from 90 Hz to 11.3 kHz, and to provide active noise control (ANC) from 50 to 800 Hz to complement the passive attenuation of the HPD. The task involves optimizing the speech SNR for each communication channel subband, subject to limiting the maximum sound level at the ear, maintaining a speech SNR preferred by users, and reducing large inter-band gain differences to improve speech quality. The performance of a proof-of-concept device has been evaluated in a pseudo-diffuse sound field when worn by human subjects under conditions of environmental noise and speech that do not pose a risk to hearing, and by simulation for other conditions. For the environmental noises employed in this study, subband speech SNR control combined with subband ANC produced greater improvement in word scores than subband ANC alone, and improved the consistency of word scores across subjects. The simulation employed a subject-specific linear model, and predicted that word scores are maintained in excess of 90% for sound levels outside the HPD of up to ∼115 dBA.

This paper presents a methodology for implementing an active headset by using H infinity robust control theory. The adopted structure is feedback tracking control. Performance, stability, and robustness of the closed-loop system have been taken into account in the design procedure by using a general framework of the H infinity theory. The resultant controller is realized on the basis of operational amplifier circuitry. Experiments are conducted to test the developed headset. The result shows that the headset achieves broadband attenuation up to approximately 15 dB in the band 200-800 Hz. The design considerations indicated in the experimental result are also addressed.

The basic idea of active noise cancellation (ANC) is to generate a controllable secondary source which compensates a noise in a certain spatial region. In headset applications, it is desirable to generate a secondary source which cancels the acoustic noise field, while preserving the useful audio field. This requires that the cancellation algorithm possess an ability to differentiate between the useful audio signal and the cancellation residual at the secondary‐sensor output. This is a formidable task for the conventional linear time‐invariant filtering approach, since the Fourier spectra of acoustic noise and the audio signal occupy the same frequency bandwidth. In this paper, a two‐sensor ANC scheme is presented with application to a noise‐reduction headset. The active noise cancellation carried out is based on the concept of the separation of superimposed signals with complete overlapping spectra by using an almost‐symmetrical time‐varying autoregressive‐moving average model (ASTV‐ARMA). The significan...

During the Apollo Program, NASA recorded dozens of audio channels to document the coordination activity in the Apollo Mission Control Center. We have digitized these recordings for the entire Apollo 11 mission, but several challenges arise when seeking to use these unique recordings for mission reconstruction. One is that 50 of the 56 recorded channels contain headset audio that documents the mix of radio and intercommunication “loops” that an individual flight controller actually heard. Reconstructing the audio that was available for selection therefore requires multi-channel source separation. The same loop might be recorded on multiple time-synchronized channels, and the some known speakers routinely speak only on certain loops, so a considerable amount of side information is available for this source separation task. Second, several additional audio channels are available from different recorders, notably including recordings made aboard the two Apollo spacecraft). Integrating these additional channel...

The perception of rooms involves different modalities, particularly hearing and seeing. Fundamental questions of audio-visual room perception have, however, not been answered yet. We investigated to what extent aesthetic judgments on music and speech performances in different spaces are based on hearing, seeing, and their interaction. Meeting methodological criteria such as optoacoustic commensurability and optoacoustic dissociation, and applying a data-based, three-dimensional, high-resolution optoacoustic simulation providing nearly all perceptually relevant physical cues allowed for a valid proportionate quantification of these shares for the first time. BRIRs and panoramic stereoscopic images were acquired in six rooms. Recordings of both a music and a speech performance were put in the rooms by applying dynamic binaural synthesis and chroma-key compositing. The scenes were played back by the use of a linearized extraaural headset and a semi-panoramic stereoscopic projection system. Test participants ...

A teleconference system was developed to transmit speech signals for audio communication with remote users. The hardware system was composed of 3-D microphone arrays to capture directional sound and a wireless headset to provide freedom of movement for users. An application programming interface (API) was used for interaction between the computer and the audio devices, including the microphone, headphone, and ADDA converter, transferring input/output audio signals in real time through the network. For the evaluation of the system, a latency test was performed with a simultaneous recording system in capturing and reproduction positions. The latency from the network procedure was calculated with different buffer sizes, and the lower-delay and smaller-deviation buffer was selected. In addition, the transmitted audio signals were compared with the input signals in terms of signal patterns and frequency responses. The resulting audio qualities in the time and frequency domain were suitable for a teleconferencing system.

We evaluate a variety of audio recording techniques for a project on the automatic analysis of speech dialog in middle school and high school classrooms. In our scenario, the teacher wears a headset microphone or a lapel microphone. A second microphone is then used to collect speech and related sounds from students in the classroom. Various boundary microphones, omni-directional microphones, and cardioid microphones are tested as this second classroom microphone. A commercial microphone array [Microsoft Xbox Kinect] is also tested. We report on how well digital source-separation techniques work for segregating the teacher and student speech signals from one another based on these various microphones and placements. We also test the recordings using various automatic speech recognition engines for word recognition error rates under different levels of background noise. Preliminary results indicate one boundary microphone, the Crown PZM-30, to be superior for the classroom recordings. This is based on its performance at capturing near and distant student signals for ASR in noisy conditions, as measured by ASR error rates across different ASR engines.

This briefing will describe and demonstrate fluidic-based sound reception, amplification, and pneumatic headphone presentation to the listener. Fluidics is non-electronic and uses laminar proportional amplifiers and fluid-flow technology to demonstrate a sensitive “microphone” due to its massless diaphragm (laminar jet of low-velocity air) and perfect impedance matching to the propagating medium (also air). Many years ago the U.S. Army developed a system called the Individual Soldier Operated Personal Acoustic Detection System (ISOPADS) that utilized this fluidic technology to enhance and extend the Soldier's listening range. This demonstration system will let people point a small parabolic dish and listen around the room using a pneumatic headset. It makes for a fun demo, especially when participants understand that they are listening to a non-electronic system that uses only air.

A detailed analysis of the acoustic transfer functions has been investigated for different locations of the headset microphones in the area of human head. Amplitude-frequency and phase-frequency characteristics of acoustic signals are discussed in the report. The signals have been recorded with the micro-mechanical-electrical-system (MEMS) microphones located on a mannequin, simulating the acoustic properties of the human torso. The the source of the signals is an emitter built in the mouth of the manikin's head. Analysis of the characteristics has shown that the acoustic signals undergo significant phase and amplitude changes associated with the interference and diffraction processes. These changes result in significant nonlinearity phase characteristics, and this nonlinearity depends on the installation location of the MEMS microphone. Also quality of speech intelligibility has been estimated by the method of expert evaluation and the speech has been recorded for different positions of the microphones and speaker head. A comparative analysis of the results has shown the optimal location of the microphones in the ear and on the breast of man.

Acoustical voice-feedback systems include headsets connected through a connector or central hub for vocalizing and sound formation assistance. A user's voice is conveyed directly into his or her own ear as well as the ears of each connected user. A central hub allows selective connection of one or more headsets to each other, or allows an instructor to selectively connect an instructor headset to one or more instructee headsets.

On January 27, 1933, Paul Lueg submitted his patent application for an active noise control system. In the intervening 80 years, the concept has progressed from simple single-input single-output feedback control systems through to complex MIMO approaches implementing a seemingly endless variety of control algorithms. Academic publications appear to continue at approximately a constant pace over the past decade, with much attention paid to algorithm refinement and development of applications. Commercial successes of active noise control include ANC headset in the aviation and consumer markets, as well as systems installed in passenger vehicle (some of which also incorporating active vibration control systems). This paper will provide a brief historical retrospective on the development of active noise control, and survey the current state of academic research as well as existing and proposed production applications (consumer, aviation, defense, etc.).

A masked-threshold and a loudness-balance method have been developed to estimate the attenuation of communication headsets and hearing protectors with built-in active noise reduction (ANR) systems. Both methods are used to estimate the attenuation of the ANR systems and the masked-threshold method is also used to estimate the total attenuation (active plus passive) of the device. The procedures are designed to be used in the presence of environmental noise, and to minimize the noise exposure of subjects during the measurements. For comparison, physical measurements of insertion loss have also been performed using a miniature microphone in the concha. Experiments showed that the masked-threshold methods tends to give increased estimates of the attenuation if the noise reduction of the left and right earcup ANR systems differs, as commonly occurs in practice. In contrast, the loudness-balance method reduces the estimates of the active attenuation. Insertion loss measurements may be influenced by the position of the microphone, owing to the spatial variability of the sound field under an earmuff when the ANR system is operating. Differences between physical and subjective measurements of up to 20 dB have been obtained in this study at frequencies of 250 Hz and below for a device in which the sound pressure varied substantially near, and within, the ear canal.

Military signal operators listen, transcribe, and respond to audio traffic over multiple audio channels, in high-level noise from vehicles and weapons. The messages typically overlap in time and may be difficult to disentangle. Two studies were carried out to determine the benefit of supplemental texting. Normal-hearing participants were tested in a mock up of a military command post. Brief messages were played simultaneously over a communications headset (dichotic) and a loudspeaker array, either in quiet or in a background of vehicle noise. The at-ear speech-to-noise ratio was 5 dB. Only those messages beginning with a pre-assigned call sign were encoded. Mean scores of 84% or better were observed for messages presented over the headset, although there was a clear right ear advantage in noise. Messages coming over the loudspeakers were more difficult to understand but a visual cue directing attention to the source of an incoming targeted message resulted in a significant improvement of 7%. Replacing audio messages over the loudspeakers in noise with visual or audiovisual presentations resulted in an improvement from 71% to 96% that did not negatively affect headset performance. The data suggest that texting is a viable option for communication in cases of degraded audio.

A headset includes a housing having an inner side thereof engageable at least with a human ear and a curved member for attaching the housing behind the ear. An audible sound emitting speaker is mounted within the housing and a microphone capable of receiving sound wave and for converting the sound wave into an electrical signal is connected to one end of a boom. The other end of the boom is attached to the housing. An amplifier is disposed within the housing and is operatively connected to the speaker and the microphone for amplifying the electrical signal. A manually operable variable volume control resistive switch is operatively disposed between the amplifier and the microphone for controlling amplification of the electrical signal. A battery is contained within the housing and supplies a predetermined voltage to the amplifier and microphone. A manually operable switch controls voltage supply from the battery.

A vibrating headset capable of generating a vibration of variable operational characteristics in the earpiece of the vibrating headset is disclosed. The vibrating headset may include one or more earpieces, each of which includes a speaker and a vibrator mechanism, wherein the vibrator mechanism receives a vibration input signal which is distinct from the audio input signal received by the speaker and defines the variable operational characteristics of the vibration to be generated by the vibrator mechanism. A gaming apparatus including a gaming device and a vibrating headset communicably coupled to the gaming device for generating vibrations of variable operational characteristics in response to receiving a vibration input signal from the gaming device is also provided. Finally, a method of creating vibrations defined by at least one variable operational characteristic in an earpiece of a vibrating headset is provided.

Systems and methods for operating a telecommunications device in whisper mode are presented. The method generally includes adjusting a transmit audio signal gain responsive to a transmit audio signal voice activity status, transmit audio signal speech level, transmit audio signal signal-to-noise ratio, and receive audio signal voice activity status. A sidetone feedback signal gain is adjusted in conjunction with adjusting the transmit audio signal gain.

A method of limiting the maximum permitted sound volume in an earphone, a headset or the like, includes the step that the size of the driving signal for driving the earphone is sensed. The driving signal has a voltage that is less than approx. 0.7 V. If the driving signal exceeds a certain level, a part of the driving signal is shunted past the earphone. A second part of the driving signal is diverted. Either the voltage of the diverted part is increased and controls the shunt device by which the first part of the driving signal is shunted past the earphone, or it is supplied to a unit with a detector device and the shunt device. The unit is sensitive to work at voltages below approx. 0.7 V. An earphone or the like for the above-mentioned method comprises a shunt device and a detector device whose current source is the signal source of the earphone.

Disclosed are methods and devices for communicating between a user device, such as a mobile terminal, and a three wire headset assembly having two microphones coupled thereto. The methods provide two microphone uplink communication and stereo audio playback on a three wire headset assembly without requiring a fourth wire or the addition of a separate voice processor. The headset assembly includes a primary microphone and secondary microphone spaced some distance apart for applying techniques of noise reduction and suppression. The headset assembly includes two speakers. Controlled switches are also included in both the user device and headset assembly to achieve dual use for a single wire of the three wires. The three wire headset may advantageously support both stereo playback and two microphone voice processing. In some embodiments, two microphone noise suppression for a wired headset mode of the user device is provided.

A new type of headset that employs adaptive noise suppression, multiple microphones, a voice activity detection (VAD) device, and unique mechanisms to position it correctly on either ear for use with phones, computers, and wired or wireless connections of any kind is described. In various embodiments, the headset employs combinations of new technologies and mechanisms to provide the user a unique communications experience.

An apparatus and method are provided for determining the donned or doffed state of a headset. In one example, a headset includes a processor, an acoustic transducer, and a detector operably coupled to the processor, the detector providing an output charge pattern corresponding to a state selected from the group consisting of the headset being donned and doffed. Advantageously, the present disclosure provides for reliably determining a donned or doffed state of a headset for efficiently routing calls, text messages, and/or otherwise being used for notifications and requests in a system.

A wireless device for use with speech recognition applications comprises a frame generator for generating successive frames from digitized original audio signals, the frames representing portions of the digitized audio signals. An autocorrelation circuit generates a set of coefficients for each frame, the coefficient set being reflective of spectral characteristics of the audio signal portion represented by the frame. In one embodiment, the autocorrelation coefficients may be used to predict the original audio signal to be subtracted from the original audio signals and to generate residual signals A Bluetooth transceiver is configured for transmitting the set of coefficients and/or residual signals as data to another device, which utilizes the coefficients for speech applications.

This includes systems and methods for noise cancellation and power management in a wireless headset. The wireless headset can perform noise cancellation by using two or more omni-directional microphones to generate a noise canceling “cone.” Audio signals received outside of the cone can be filtered out. An accelerometer can be used to determine the location of the wireless headset and the system can then properly position the cone to face the user's voice and filter out other audio signals. Additionally, by monitoring the ambient noise, the system can save power by turning off the noise cancellation system when the amount of ambient noise is below a threshold value.

Electronic devices and accessories for electronic devices such as headsets are provided. The electronic devices may produce audio output. The headsets may include earbuds with speakers that play the audio output for a user while the earbuds are located in the user's ears. Circuitry in an electronic device and a headset may be used in evaluating how well the earbuds are sealed to the user's ears. In response to seal quality measurements, informative messages can be generated for the user, overall earbud volume may be increased, balance adjustments may be made to correct for mismatched balance between left and right earbuds, equalization settings may be adjusted, and noise cancellation circuitry settings can be changed. Electrical impedance measurements and acoustic measurements can be used in evaluating seal quality.

A digital headset system for use in audiometric testing. The headset system includes a stored calibration reference relating the exact frequency and volume response of each speaker to analog input signals. A microprocessor accesses the calibration reference to determine the required analog signal needed to produce the desired sound. An on-board digital to analog converter generates the required analog signal and transmits it to the speaker. The headset is used with software based audiometric test methods that allow the generation of an electronic user hearing profile. The user hearing profile may be used to re-program an audio emitting device to function in a customized manner. The user hearing profile may be overlaid on a partner hearing profile in order to emulate for the partner how sounds are perceived by the user. The test methods may be self-administered or administered by a healthcare professional.

Active noise cancellation (ANC) technologies have been successfully applied in headsets to reduce ambient noise in the ear canal. The performance of such a technology depends on the characteristics of the secondary path (SP). In this paper, two different systems are studied: one system utilizes a moving-coil speaker as the secondary source and forms a closed cavity over the ear, while the other one uses a balanced-armature receiver inside the ear canal as the secondary source. The differences in the SP responses of these two systems are investigated both experimentally and theoretically. The implications for designing an effective ANC solution will be discussed based on numerical simulations.

Magnetic resonance imaging (MRI) is extensively used in clinical and medical researches because MRI is non-invasive and non-radiation. However, using the MRI cause the loud acoustic noise generated by gradient switching. The MRI noise is annoying patients. Therefore, active noise cancellation (ANC) is used to solve this problem in this article. In this article, the main work is to implement real-time system using DSP. The noise generated by echo planar imaging (EPI), which is generally used in fMRI researches, is recorded from 3 Tesla MRI for testing ANC system. The EPI signal is regular and predictable, means the signal like a pattern, we using LMS algorithm to training the signal and learning predict the noise spectrum. After the calculated the noise, we output the reverse single to we home-built headset.

A number of projects have been carried out in Nanjing University to implement active noise control technologies, which include active control of transformer noise, active sound barrier, active noise control in communication chassis, active noise control in natural ventilation windows, active control of large impulsive noise in headset and active noise control in a train compartment. Unlike our previous research, these projects are all funded by industries and the aim is not for doing academic research, but to make commercial prototypes. Various challenges in the implementation of active noise control technologies in these projects to make commercial products are reported and discussed, and main issues to make successful commercial active noise control products are pointed out.

In an audio headset having one or more far-field microphones mounted to the headset; and one or more speakers mounted to the headset environmental sound may be recorded using the one or more far-field microphones and mixed with source media sound to produce a mixed sound. The mixed sound may then be played over the one or more speakers.

Apparatus and methods are provided for enabling a microphone module that connects to a headset jack on a game controller. The microphone may include directional or omni-directional microphone element(s) that pick up voice input from people near a peripheral, such as a game controller, but reject background sounds from the room. User controls may also be available to set the microphone gain level, either manually or automatically, and/or to mute or unmute the microphone. The microphone of the invention can be used on a wired or wireless game controller. When used with a wireless game controller, wireless microphone capabilities and freedom of movement are provided without headdress requirements imposed by headsets.

A method of converting electrical signals into mechanical vibration by means of a transducer in the human ear, an ear canal transducer and a headset wherein a sensation of hearing is achieved by exciting the tissue of the ear canal directly with said transducer, whereby the vibrations propagate to the tympanic membrane and into the human sound sensing organs.

A visual feedback system (101) that activates a visual display (102) when the sound pressure level from a headset (105) attached to the system exceeds a preset level is provided, along with a method of using the same. The visual feedback system is interposed between the audio source (103) and the headset (105) and is either integral to a specific headset or coupleable to any of a variety of headsets. If a non-integral headset is used with the visual feedback system, the system is matched to the characteristics of the selected headset, for example using a selector switch or via a calibration process. During operation, the visual feedback system illuminates a display (e.g., an LED) whenever the sound pressure level from the attached headset exceeds the preset level. The visual feedback system can be implemented using analog or digital circuitry.

A headset is disclosed. The headset includes an earpiece, a voice conducting and connecting system and a microphone connected to the voice conducting and connecting system, the microphone being capable of operating in a directional configuration and in an omnidirectional configuration.

A communication device includes: (1) a wireless adapter at which a wireless headset is communicatively connected to the communication device and at which is received a first acoustic input that includes a speech input and a first ambient noise input; (2) a microphone that receives a second acoustic input, which includes a second ambient noise input; and (3) a dual-channel adaptive noise canceller that utilizes the second ambient noise input to filter the first ambient noise input out of the first acoustic input to generate an acoustic output that primarily comprises the speech input.

A headset apparatus is taught comprising at least one sound emission means, a tube connected to the sound emission means adjacent one end of the tube, and an outlet at the opposite end of the tube for positioning the outlet at a desired position adjacent a user's ear, various positions possible to achieve a variety of acoustic effects. The apparatus may comprise two or four sound emission means, with a plurality of tubes connecting the sound emission means to outlets, to effect diverse directional acoustic effects. Electronic control means may be employed to control emissions of the sound emission means, and the apparatus may comprise bass sound emission means adjacent the user's ear. Supporting means are also taught for supporting the headset apparatus on a user, such that the user's ear can be moved relative to the headset apparatus to enhance the acoustic effects.

Many workers refuse to wear hearing protection devices (HPDs) because they would rather accept the health risks than sacrifice the ability to communicate with coworkers. Integrating active noise reduction (ANR) techniques with speech enhancement algorithms could solve these limitations of modern electronic HPDs. An adaptive delay feedforward subband structure has been implemented by forming parallel signal filtering and filter update paths for each frequency band. Subband ANR provides additional attenuation of environmental noise beyond that of passive HPDs within the lower frequencies associated with speech communication. The subband structure can also provide valuable information regarding the spectral content of the environmental noise that can be exploited to determine where additional power is needed in a communication channel to improve intelligibility. The system, initially developed in simulation, maintains a specified signal-to-noise ratio in each subband while simultaneously limiting the power added by the communication channel to below that associated with hearing damage. Speech Transmission Index models have been used to validate the improvements in speech intelligibility over typical passive and ANR communication headset designs. Preliminary results for a circumaural HPD constructed to implement the concept will be presented for comparison with the simulation. [Work supported by NIOSH (R01 OH008669).]

Novel techniques for hearing enhancement based on a pair of glasses are disclosed. In one embodiment, the glasses include a first lens holder, a second lens holder, a bridge element, a first temple, a second temple, a first speaker, a second speaker, a microphone and at least one electrical component that performs a hearing enhanced function to enhance audio signals from the microphone to be sent to the speakers for the user to hear. At least one electrical component in the glasses is for generating other audio signals by the speakers, with the other audio signals originating from signals other than audio signals captured by the microphone. In another embodiment, some of the electrical components are in a portable device, wired or wirelessly, coupled to the glasses. In yet another embodiment, the glasses function as a headset, with other electrical components in the portable device. In such embodiment, since the glasses can provide a number of different hearing functions, a third party may not be able to tell whether the user is wearing the glasses to have his hearing enhanced, or for hearing other audio signals.

A headset that includes an input that receives an audio signal from an electronic equipment, and at least one earpiece having an electro-acoustic speaker for converting electrical signals to audible sound. The headset further includes at least one microphone that receives ambient sound and converts the ambient sound to an ambient sound signal. In addition, the headset includes a processing circuit formed as an integral unit with the input and the at least one earpiece. The processing circuit includes circuitry that combines the audio signal from the electronic equipment with the ambient sound signal from the at least one microphone and provides the combined signals to the at least one earpiece such that the electro-acoustic speaker reproduces the ambient sound in combination with audio from the audio signal.